DTMF Issue on CUCME 7.0

[gmgarrian]

I have two sites with CUCME 7.0 routers. Site A has a PRI for outbound calls and a T1 connection to Site B. Site B forwards its outbound calls to Site A to go out the PRI.

When Site B makes an outbound call the call connects, but after that it doesn't forward any DTMF tones. You can hear the tones on the handset but it doesn't forward the tones outbound.

Site A does not have this problem.

I've tried a number of different dtmf-relay configs but nothing seems to help.

Here's the current dial peer config:

!

dial-peer voice 152 voip

description <<11 Digit Long Distance Numbers>>

preference 1

destination-pattern 91[2-9]T

session protocol sipv2

session target ipv4:192.168.150.1

dtmf-relay sip-notify

codec g711ulaw

no vad

!

[Nicholas Matthews]

I would shy away from sip-notify. It's primary use is for CUE, and that's about it.

I would use rtp-nte for SIP configuration.

Otherwise you are probably hitting dial peer 0. Do a 'debug voip dialpeer'. Be aware the the outgoing dial peer on one router must have the DTMF, as well as the INCOMING dial peer on the other router. This is most likely your problem, as incoming dial peers are very commonly overlooked.

Try adding an 'incoming called-number .' to your outgoing dial peer on both sides and see if you get different results.

hth,

nick

[paolo bevilacqua · Accepted solution]

To add something to Nick's good advice.

You might not know that the "preferred" protocol for trunking CMEs is H.323. When you switch to it, certain things magically start working. Not to say that this is necessarily the case here.

[gmgarrian]

How would I go about switching to an H.323 trunk?

[Kenneth Mohammed · Accepted solution]

Hello,

You switch to H323 trunk simply by removing the "session protocol sipv2" statement from your dial peer pointing to the other CME.

Hope that helped, if so please rate.

[gmgarrian]

I have tried rtp-nte and h245-alphanumeric without any luck. I have not checked to make sure the dtmf line is also part of the incoming dial peer as well. I'll also try adding the "incoming called-number ." to the outbound dial peer. One question on that though, won't that conflict with the existing incoming dial peer for inbound calls on the PRI?

[paolo bevilacqua]

Hi, incoming called-number doesn't do anything for outgoing DP, and if trunking is working already, you don't need it at all. But, you need incoming-calling number on the incoming voip DP to make sure the dtmf relay agrees on both sides.

Also, recommend you try H.323.

[gmgarrian]

Thanks to all! I removed the "session protocol sipv2" from the outboud dial peer and it's now working.

Posts have been rated.

Thanks again!

[Kenneth Mohammed]

Greg,

Glad to hear you got it working and thanks for the nice rating!